In a mobile telephony system, ancillary information (e.g., signaling information, overhead, enhanced forward error correction channel coding) is needed to adjust, control, and coordinate the system's configuration and operation. In some instances, the need to communicate ancillary information to a far-end mobile may arise while the far-end mobile is in use. When this occurs, the mobile and the base station combine the ancillary information with voice traffic. If the bandwidth on the wireless link leading to the far-end mobile is fully occupied, the coding rate of the voice traffic will need to be reduced to make room for the ancillary information.
In another scenario, congestion in a packet network may require a rate reduction to be effected, in order to allow a call to continue to be at least minimally supported between two end points so that the call is not dropped. Such requirement for a rate reduction may occur at random times, irrespective of the coding rate of voice traffic traveling in the packet network.
To achieve rate reduction in a network that carries packets of coded voice traffic, several methods have been proposed. One rather rudimentary way of effecting rate reduction of coded voice traffic traveling in a packet network is to drop packets. In this mode of operation, a packet (or plural packets) of coded voice traffic is/are suppressed (i.e., not transmitted, or “blanked”) in order to liberate bandwidth, either downstream in the packet network or on the wireless link with the far-end mobile. However, the consequence of such drastic deletion of packets is a degradation of the recovered speech that could lead to a severe loss of intelligibility.
A slightly more sophisticated multiplexing technique for rate reduction of coded voice traffic traveling in a packet network consists of decoding (i.e., synthesizing) a received packet of coded voice traffic that was coded at an original (i.e., higher) rate. The fully synthesized speech signal is then re-coded at a lower rate, thereby preserving certain characteristics of the original speech, while freeing up bandwidth to insert the ancillary information or to alleviate network congestion. The operation of decoding the coded voice traffic into recovered speech and re-coding the recovered speech at a different (i.e., lower) rate is known as transcoding (or “tandem operation”), which has the disadvantage of requiring the processing and memory resources for a full codec just to provide rate reduction functionality. In the case of most codecs, the additional resources/cost associated with providing rate reduction functionality of the type described above are considered too high for mass implementation. In addition, transcoding exposes the speech to possible degradation as it is synthesized and then re-coded.
Moreover, both of the above techniques can lead to severe degradations in voice quality during prolonged periods of a required rate reduction, such as may occur when, for example, two air interfaces need to run at different packet rates for a mobile-to-mobile call. In such cases, the coded voice traffic emanating from the near-end mobile may need to be reduced by the network before being transmitted to the far-end mobile until the radio condition improves. Such a situation may last for several seconds or even minutes, which tends to have significant deleterious effects on intelligibility when conventional rate reduction methods are employed.
Therefore, a need exists in the industry to provide an improved mechanism for reducing the coding rate of coded voice traffic traveling in a packet network without significantly affecting voice quality.